EP3685378A1 - Signal processor and method for providing a processed audio signal reducing noise and reverberation - Google Patents

Abstract

A signal processor for providing one or more processed audio signals based on one or more input audio signals, wherein said signal processor is configured to estimate coefficients of an autoregressive reverb model to using input audio signals and delayed noise reduced reverberated signals obtained by noise reduction. The signal processor is configured to provide reduced noise reverberated signals using the input audio signals and estimated coefficients of the autoregressive reverb model. The signal processor is configured to derive reduced noise, reduced reverberation output signals using the reduced noise reverberated signals and estimated coefficients of the autoregressive reverberation model. The present invention also relates to a method and a computer program comprising a similar functionality.

Description

Signal Processor and Method for Providing a Processed Audio Signal Reducing

Noise and Reverberation

Description

Technical Field

Embodiments according to the invention are related to a signal processor for providing a processed audio signal.

Further embodiments according to the invention are related to a method for providing a processed audio signal. Further embodiments according to the invention are related to a computer program for performing said methods.

Embodiments according to the invention are related to a method and apparatus for online dereverberation and noise reduction (for example, using a parallel structure) with reduction control.

Further embodiments according to the invention are related to linear prediction based online dereverberation and noise reduction using alternating Kalman filters. Embodiments according to the invention relate to a signal processor, a method and a computer program for noise reduction and reverberation reduction.

Background of the Invention Audio signal processing, speech communication and audio transmission are continuously developing technical fields. However, when handling audio signals, it is often found that noise and reverberation degrade the audio quality.

For example, in distant speech communication scenarios, where the desired speech source is far from the capturing device, the speech quality and intelligibility is typically degraded due to high levels of reverberation and noise compared to the desired speech level.

Also the performance of speech recognizers degrades drastically in distant talking scenarios [15], [34].

Therefore, dereverberation in noisy environments for real-time frame-by-frame processing with high perceptual quality remains a challenging and partly unsolved task. State-of-the-art multichannel dereverberation algorithms are based on spatio-spectral filtering [2], [27], system identification [25], [26], acoustic channel inversion [20], [22] or linear prediction using an autoregressive (AR) reverberation model [21], [29], [32]. Successful application of the linear prediction based approaches was achieved by using a multichannel autoregressive (MAR) model for each short-time Fourier transform (STFT) domain frequency band. Advantages of methods based on the MAR model are that they are valid for multiple sources, they directly estimate a dereverberation filter of finite length, the required filters are relatively short, and they are suitable as pre-processing techniques for beamforming algorithms. A great challenge of the MAR signal model is the integration of additive noise, which has to be removed in advance [30], [32] without destroying the relations between neighboring time-frames of the reverberant signal. In [33], a generalized framework for the multichannel linear prediction methods called blind impulse response shortening was presented, which aims at shortening the reverberant tail in each microphone and results in the same number of output as input channels, while preserving the inter-microphone correlation of the desired signal.

As the first solutions based on the multichannel linear prediction framework were batch algorithms, further efforts have been made to develop online algorithms, which are suitable for real-time processing [4, 12, 13,31 ,35]. However, the reduction of additive noise in an online solution has been considered only in [31 ] to the best of our knowledge.

In view of the conventional solutions, there is a desire for a concept which provides an improved tradeoff between complexity, stability and signal quality when reducing both noise and reverberation of an audio signal.

Summary of the Invention An embodiment according to the invention creates a signal processor for providing a processed audio signal (for example, a noise-reduced and reverberation-reduced audio signal, which may be a single-channel audio signal or a multi-channel audio signal) (or generally speaking, one or more processed audio signals) on the basis of an input audio signal (for example, a single-channel or a multi-channel input audio signal) (or generally speaking, on the basis of one or more input audio signals). The signal processor is configured to estimate coefficients of an (for example, multi-channel) autoregressive reverberation model (for example, AR coefficients or MAR coefficients) using the input audio signal (for example, the noisy and reverberant input audio signal or multiple noisy and reverberant input audio signals, or directly an observed signal y(n) which may, for example, originate from one or more microphones) (or, generally speaking, using one or more input audio signals) and (one or more) delayed noise-reduced reverberant signals obtained using a noise reduction (or a noise reduction stage). For example, the delayed noise-reduced reverberant signal may comprise (one or more) past noise-reduced reverberant signals which may be represented by x( 7). For example, the estimation of the coefficients may be performed by an AR coefficient estimation stage or by an MAR coefficient estimation stage of the signal processor.

Moreover, the signal processor is configured to provide a noise-reduced reverberant signal (for example, of a current frame) (or, generally speaking, one or more noise- reduced reverberant signals) using the input audio signal (which may, for example, be a noisy and reverberant input audio signal or which may, for example, be the noisy observed signal y(n) which may originate from one or more microphones) and the estimated coefficients of the autoregressive reverberation model (which may be a multi- channel autoregressive reverberation model) (and wherein the estimated coefficients may, for example, be associated with the current frame and may, for example, be called "MAR coefficients"). Moreover, the part of the signal processor configured to provide the noise- reduced reverberant signal may be considered as a "noise reduction stage". Moreover, the audio signal processor is configured to provide a noise-reduced and reverberation-reduced output signal (or, generally speaking, one or more noise-reduced and reverberation-reduced output signals) using the noise-reduced (reverberant) signal (or, generally speaking, one or more noise-reduced, reverberant signals) and the estimated coefficients of the autoregressive reverberation model (or multi-channel autoregressive reverberation model). This may, for example, be performed using a reverberation estimation and a signal subtraction. This embodiment according to the invention is based on the finding that it is possible to overcome a causality problem, which is found in some conventional solutions, by estimating the coefficients of the autoregressive reverberation model associated with a certain frame on the basis of a delayed and noise reduced reverberant signal which may be associated with one or more preceding frames, and that it is possible to provide the noise reduced reverberant signal of the current frame using the input audio signal and the estimated coefficients of the autoregressive reverberation model associated with the current frame and obtained on the basis of noise-reduced (and typically reverberant) signals (for example, provided by the noise reduction stage) associated with one or more preceding frames. Accordingly, the computational complexity can be kept reasonably small, since the estimation of the coefficients of the autoregressive reverberation model and the estimation of the noise-reduced reverberant signal can be performed separately and alternatingly. In other words, the separate estimation of the coefficients of the autoregressive reverberation model and of the noise-reduced reverberant signal can be performed more efficiently than a joint estimation of coefficients of an autoregressive reverberation model and of a noise-reduced reverberant signal, and also more efficiently than a joint (one-step) estimation of a noise-reduced and reverberation-reduced audio signal. Nevertheless, it has been found that the consideration of delayed (or, equivalently, past) noise-reduced reverberant signals obtained using a noise reduction in the estimation of the coefficients of the autoregressive reverberation model results in a reasonably good estimation of the coefficients of the autoregressive reverberation model, such that there is no severe degradation of the audio quality of the processed signal (output signal). Accordingly, it is possible to alternatingly estimate coefficients of the autoregressive reverberation model and frames of the noise reduced reverberant signal while still obtaining a good audio quality.

Consequently, the tradeoff between complexity, stability and signal quality can be considered as good.

In a preferred embodiment, the signal processor is configured to estimate coefficients of a multi-channel autoregressive reverberation model. It has been found that the concept described herein is well-suited for a handling of multi-channel signals and brings along particular improvements of the complexity for such multi-channel signals. In a preferred embodiment, the signal processor is configured to use estimated coefficients of the autoregressive reverberation model associated with a currently processed portion (for example, a time-frame having a frame index n) of the input audio signal in order to produce the noise-reduced reverberant signal associated with the currently processed portion (for example, a time-frame having frame index n) of the input audio signal. Accordingly, the provision of the noise-reduced reverberant signal associated with the currently processed portion may rely on the previous estimation of the coefficients of the autoregressive reverberation model associated with the currently processed portion of the input audio signal, or the estimation of the coefficients of the autoregressive reverberation model associated with a currently processed portion (or frame) may precede the provision of the noise-reduced reverberant signal associated with the currently processed portion (or frame). Accordingly, when processing an audio frame with frame index n, the estimation of the coefficients of the autoregressive reverberation model may be performed first (for example, using a past noise reduced but reverberant signal) and the provision of the noise-reduced reverberant signal associated with the currently processed frame may be performed then. It has been found that such an order of the processing results in particularly good results, while a reverse order will typically not perform quite as good. In a preferred embodiment, the signal processor is configured to use one or more delayed noise-reduced reverberant signals (or, alternatively, a noise-reduced reverberant signal) associated with (or based on) a previously processed portion (for example, a frame having frame index n-1 ) of the input audio signal (for example, an input signal y(n)) for an estimation of coefficients of the autoregressive reverberation model associated with the currently processed portion (for example, having a frame index n) of the input audio signal. By using a noise-reduced reverberant signal associated with the previously processed portion (or frame) of the input audio signal for an estimation of a coefficient of the autoregressive reverberation model associated with a currently processed portion (or frame) of the input audio signal, a causality problem can be avoided, since the provision of the noise-reduced reverberant signal associated with the previously processed frame can typically be provided before the estimation of the coefficients of the autoregressive reverberation model associated with the currently processed portion (or frame) of the input audio signal. Also, it has been found that the usage of a noise reduced reverberant signal associated with a previously processed portion of the input audio signal results in a sufficiently good estimation of the coefficients of the autoregressive reverberation model. In a preferred embodiment, the signal processor is configured to alternatingly provide estimated coefficients of the autoregressive reverberation model (or multi-channel autoregressive reverberation model) and noise-reduced reverberant signal portions. Moreover, the signal processor is configured to use estimated coefficients (or, alternatively, previously estimated coefficients) of the (preferably multi-channel) autoregressive reverberation model for the provision of the noise-reduced reverberant signal portions. Moreover, the signal processor is configured to use one or more delayed noise-reduced reverberant signals (or, alternatively, previously provided noise reduced reverberant signal portions) for the estimation of coefficients of the multi-channel autoregressive reverberation model. By performing such an alternating provision of estimated coefficients of the autoregressive reverberation model and of noise-reduced reverberant signal portions, the computational complexity can be kept low and results can still be obtained with little delay. Also, computational instabilities, which could be caused by a joint estimation of coefficients of the multi-channel autoregressive reverberation model and noise reduced reverberant signal portions can be avoided.

In a preferred embodiment, the signal processor may be configured to apply an algorithm minimizing a cost function (for example, a Kalman filter, a recursive least squares filter or a normalized least mean squares (NLMS) filter) in order to estimate the coefficients of the (preferably multi-channel) autoregressive reverberation model. It has been found that usage of such algorithms is well-suited for estimating the coefficients of the autoregressive reverberation model. The cost function may, for example be defined as shown in equation (15), and the minimization may, for example, fulfill the functionality as shown in equation (17) or minimize the trace of an error matrix, as shown in equation (19). The Minimization of the cost function may, for example, follow equations (20) to (25). The minimization of the cost function may also use steps 4 to 6 of Algorithm 1.

In a preferred embodiment, the cost function used for the estimation of the coefficients of the autoregressive reverberation model (for example, in the algorithm that minimizes a cost function) is an expectation value for a mean squared error of the coefficients of the autoregressive reverberation model, for example, as shown in equation (19). Accordingly, coefficients of the autoregressive reverberation model which are expected to fit well an acoustic environment causing the reverberation can be achieved. It should be noted that expected statistical properties of the MAR coefficient noise and of the noisy dereverberated signals (state and observation noises), for example, be estimated in a separate, preparatory step (for example, using one or more of equations (26) to (29). In a preferred embodiment, the signal processor may be configured to apply the algorithm for the minimization of the cost function in order to estimate the coefficients of the (preferably multi-channel) autoregressive reverberation model under the assumption that the noise-reduced reverberant signal is fixed (for example, not affected by the coefficients of the autoregressive reverberation model associated with the currently processed portion of the input audio signal). By making such an assumption, the computational complexity can be reduced significantly and instabilities of the computation can also be avoided. For example, the algorithm of equations (20) to (25) makes such an assumption.

In a preferred embodiment, the signal processor is configured to apply an algorithm for a minimization of a cost function (for example, a Kalman filter or a recursive least squares filter or a NLMS filter) in order to estimate the noise-reduced reverberant signal. The cost function may, for example be defined as shown in equation (16), and the minimization may, for example, fulfill the functionality as shown in equation (18) or minimize the trace of an error matrix, as shown in equation (30). The minimization of the cost function may, for example, follow equations (31 ) to (36).

In a preferred embodiment, the signal processor is configured to apply an algorithm for a minimization of a cost function (for example, a Kalman filter , a recursive least squares filter or a NLMS filter) in order to estimate the noise-reduced reverberant signal. It has been found that the usage of such an algorithm for a minimization of a cost function is also very efficient for the determination of the noise-reduced reverberant signal, for example, if statistical properties of the noise are known or estimated. Moreover, the computational complexity can be substantially improved if similar algorithms (for example, algorithms minimizing a cost function) are used both for the estimation of the coefficients of the autoregressive reverberation model and for the estimation of the noise-reduced reverberant signal. For example, the algorithm according to equations (31 ) to (36) may be used, wherein parameters to be used in said algorithm may be determined according to one or more of equations (37) to (42). Also, the functionality may be performed using steps 7 to 9 of Algorithm 1 .

In a preferred embodiment, the cost function used for the estimation of the (optionally noise-reduced) reverberant signal is an expectation value for a mean-squared error of the (optionally noise-reduced) reverberant signal. It has been found that such a cost function (for example, according to equation (16) or according to equation (30)) provides for good results and can be evaluated using reasonable computational effort. Moreover, it should be noted that the estimation of the mean squared error of the noise-reduced reverberant signal is possible, for example, if information (or assumption) regarding statistical characteristics of the noise (for example, the noise covariance matrix) and possibly also regarding the desired signal (for example, the desired speech covariance matrix) are available.

In a preferred embodiment, the signal processor is configured to apply the algorithm for the minimization of the cost function in order to estimate the (optionally noise-reduced) reverberant signal under the assumption that the coefficients of the autoregressive reverberation model are fixed (for example, not affected by the noise-reduced reverberant signal associated with the currently processed portion of the input audio signal). It has been found that such an "ideal" assumption (which is, for example, made in the computation according to equations (31 ) to (36)) does not significantly degrade the results of the estimation of the noise-reduced reverberant signal but significantly reduces the computational effort (for example, when compared to a joint estimation of the noise- reduced reverberant signal and the coefficients of the autoregressive reverberation model, or when compared to a direct estimation of a noise-reduced and reverberation-reduced output signal (in a single-step procedure)).

Furthermore, the assumption allows for an alternating procedure in which the noise- reduced reverberant signal and the coefficients of the autoregressive reverberation model are estimated in a separated manner (for example, by alternatingly performing steps 4 to 6 and steps 7 to 9 of Algorithm 1 ).

In a preferred embodiment, the signal processor is configured to determine a reverberation component on the basis of estimated coefficients of the (preferably multichannel) autoregressive reverberation model and on the basis of one or more delayed noise-reduced reverberant signals (or, alternatively, on the basis of the noise-reduced reverberant signal) associated with a previously processed portion (for example, a frame) of the input audio signal (for example, by filtering the noise-reduced reverberant signal using the estimated coefficients of the autoregressive reverberation model). Moreover, the signal processor is preferably configured to (at least partially) cancel (for example, subtract) the reverberation component from the noise-reduced reverberant signal associated with a currently processed portion (for example, a frame) of the input audio In a preferred embodiment, the signal processor is configured to estimate a statistic (for example, a covariance) (or a statistical property) of a noise component of the input audio signal. Such a statistic of the noise component of the input audio signal may, for example, be useful in the estimation (or provision) of a noise-reduced reverberant signal. Also, an estimation (or determination) of a statistic of the noise component of the input audio signal can facilitate a formulation of a cost function because the statistic of the noise component of the input audio signal can be used as a part of said cost function.

In a preferred embodiment, the signal processor is configured to estimate a statistic (for example, a covariance) (or a statistical property) of a noise component of the input audio signal during a non-speech period (wherein, for example, the non-speech period is detected using a speech detector). It has been found that a detection of non-speech periods is possible with reasonable effort and it has also been found that the noise which is present during non-speech periods is typically also present during the speech periods without too many changes. Accordingly, it is possible to efficiently obtain the statistics of the noise component, which are useable for the provision of the noise-reduced reverberant signal.

In a preferred embodiment, the signal processor is configured to estimate the coefficients of the (preferably multi-channel) autoregressive reverberation modeled using a Kalman filter. It has been found that such a Kalman filter allows for an efficient computation and is well-adapted to the requirements of the signal processing task. For example, the implementation according to equations (20) to (25) can be used. In a preferred embodiment, the signal processor is configured to estimate the coefficients of the (preferably multi-channel) autoregressive reverberation model on the basis of an estimated error matrix of a vector of coefficients of the (preferably multi-channel) autoregressive reverberation model (for example, associated with a previously processed portion of the audio signal), on the basis of an estimated covariance of an uncertainty noise of the vector of a coefficient of the (preferably multi-channel) autoregressive reverberation model (for example, as given in equation (26)), on the basis of a previous vector of (estimated) coefficients of the (preferably multi-channel) autoregressive reverberation model (for example, associated with a previously processed portion or version of the input audio signal), on the basis of one or more delayed noise-reduced reverberant signals delayed noise-reduced reverberant signals (for example, (past) noise- reduced reverberant signals, represented by X(n), for example associated with previous portions or frames of the input audio signal), (optionally) on the basis of an estimated covariance associated with noisy (for example, non-noise-reduced) but reverberation- reduced (or reverberation-free) signal components of the input audio signal, and on the basis of the input audio signal, !t has been found that estimating the coefficients of the autoregressive reverberation model on the basis of these input variables is both computationally efficient and brings along accurate estimates of the coefficients of the autoregressive reverberation model.

In a preferred embodiment, the signal processor is configured to estimate the noise- reduced reverberant signal using a Kalman filter. It has been found that usage of such a Kalman filter (which may implement the functionality as given in equations 31 to 36) is also advantageous for the estimation of the noise-reduced reverberant signal. Also, using a Kalman filter both for the estimation of the coefficient of the autoregressive reverberation model and for the estimation of the noise-reduced reverberant signal can provide good results.

In a preferred embodiment, the signal processor is configured to estimate the noise- reduced reverberant signal on the basis of an estimated error matrix of the noise-reduced reverberant signal (for example, associated with a previously-processed portion or frame of the input audio signal, for example), on the basis of an estimated covariance of a desired speech signal (for example, associated with a currently processed portion or frame of the input audio signal, for example, as given in equations 37 to 42), on the basis of one or more previous estimates of the noise-reduced reverberant signal (for example, associated with one or more previously processed portions or frames of the input audio signal), on the basis of a plurality of coefficients of the (preferably multi-channel) autoregressive reverberation model (for example, associated with the currently processed portion or frame of the input audio signal, for example defining a matrix F(n)), on the basis of an estimated noise covariance associated with the input audio signal, and on the basis of the input audio signal. It has been found that the estimation of the noise-reduced reverberant signal on the basis of these quantities is both computationally efficient and provides for a good quality of the audio signal.

In a preferred embodiment, the signal processor is configured to obtain an estimated covariance associated with noisy but reverberation-reduced (or non-reverberant) signal components of the input audio signal on the basis of a weighted combination (for example, according to equation 28) of a recursive covariance estimate determined recursively using previous estimates of noisy but reverberation-reduced (or non- reverberant) signal components of the input audio signal (for example, associated with previously processed portions or frames of the input audio signal, for example according to equation 29) and of an outer product of an (for example, intermediate) estimate of noisy but reverberation-reduced (or non-reverberant) signal components of the input audio signal (for example, associated with a currently processed portion of the input audio signal). For example, the intermediate estimate of the noisy but reverberation-reduced signal components may be obtained as an innovation in a Kalman filtering process (for example, according to equation (22)). For example, the intermediate estimate may be a prediction using predicted coefficients (for example, as determined by equation (21 )).

It has been found that such a concept provides for a good estimate of the covariance associated with noisy but reverberation-reduced (or non-reverberant) signal components with reasonable computational complexity.

In a preferred embodiment, the recursive covariance estimate of the desired signal plus noise is based on an estimation of the noisy but reverberation-reduced (or non- reverberant) signal components of the input audio signal computed using final estimate coefficients of the (preferably multi-channel) autoregressive reverberation model and using a final estimate of the noise-reduced reverberant signal (for example, according to equation (29) in combination with the definition of u(n)). Alternatively or in addition, the signal processor is configured to obtain the outer product of the noisy but reverberation- reduced signal components of the input audio signal on the basis of an intermediate estimate (for example, a prediction) of the coefficients of the (preferably multi-channel) autoregressive reverberation model (for example, in a Kalman filtering process) (for example, in order to obtain the covariance estimate)(for example obtained according to equation (21 )). By using such a concept (for example, in accordance with equations (28) and (29) described below when taken in combination with the definitions of e(n) and Ci(n)) the estimated covariance can be obtained in an efficient manner.

In a preferred embodiment, the signal processor is configured to obtain an estimated covariance associated with a noise-reduced and reverberation-reduced (or non- reverberant) signal component of the input audio signal on the basis of a weighted combination (for example, according to equation (37)) of a recursive covariance estimate determined recursively using previous estimates of a noise-reduced and reverberation- reduced signal components of the input audio signal (for example, associated with previously processed portions or frames of the input audio signal) (which may, for example, be considered as a recursive a-posteriori maximum likelihood estimate) and of an a-priori estimate of the covariance which is based on a currently processed portion of the input audio signal (and obtained, for example, in accordance with equation (41 )). In this manner, a meaningful estimate of the covariance associated with the noise-reduced and reverberation-reduced signal component of the input audio signal can be obtained with moderate computational complexity. For example, using the approach described in equation (37) allows for the usage of a Kalman filter for noise reduction with good results. In a preferred embodiment, the signal processor is configured to obtain the recursive covariance estimate based on an estimation of the noise-reduced and the reverberation- reduced (or non-reverberant) signal components of the input audio signal computed using final estimated coefficients of the (preferably multi-channel) autoregressive reverberation model and using a final estimate of the noise-reduced reverberant (output) signal (for example, using equation (38)). Alternatively or in addition, the signal processor is configured to obtain the a-priori estimate of the covariance using a Wiener filtering of the input signal (as shown, for example, in equation (41 )), wherein a Wiener filtering operation is determined in dependence on the covariance information regarding the input audio signal, in dependence on covariance information regarding a reverberation component of the input audio signal and in dependence on covariance information regarding a noise component of the input audio signal (as shown, for example, in equation (42)). It has been found that these concepts are helpful in efficient computation of the estimated covariance associated with the noise-reduced and reverberation-reduced signal component. The signal processors described here, and the signal processors defined in the claims, can be supplemented by any of the features, functionalities and details described herein, both individually and taken in combination. Details regarding the computation of different parameters can be used independently. Also details regarding individual processing steps can be used independently.

Another embodiment according to the invention creates a method for providing a processed audio signal (for example, a noise-reduced and reverberation-reduced audio signal, which may be a single-channel audio signal or a multi-channel audio signal) on the basis of an input audio signal (for example, a single-channel or multi-channel input audio signal). The method comprises estimating coefficients of a (preferably, but not necessarily, multi-channel) autoregressive reverberation model (for example, AR coefficients or MAR coefficients) using the {typically noisy and reverberant) input audio signal (or input audio signals) (for example, directly from the observed signal y(n)) and delayed (or past) noise-reduced reverberant signals obtained using a noise reduction (noise reduction stage) (for example, past noise-reduced reverberant signals x(f?)). This functionality may, for example, be performed by the AR coefficient estimation stage.

Moreover, the method comprises providing a noise-reduced reverberant signal (for example, of a current frame) using the (typically noisy and reverberant) input audio signal (for example, the noisy observed signal y(n)) and the estimated coefficients of the (preferably multi-channel) autoregressive reverberation model (for example, associated with the current frame). The estimated coefficients of the autoregressive reverberation model may, for example, be "MAR coefficients". Moreover, the functionality of providing the noise-reduced reverberant signal may, for example, be performed by a noise reduction stage.

The method further comprises deriving a noise-reduced and reverberation-reduced output signal using the noise-reduced reverberant signal and the estimated coefficients of the (preferably multi-channel) autoregressive reverberation model.

This method is based on the same considerations as the above mentioned signal processor, such that the above explanations also apply.

Moreover, the method can be supplemented by any features, functionalities and details described herein with respect to the signal processor, both individually and in combination.

Another embodiment according to the invention creates a computer program for performing the method as described herein when the computer program runs on a computer.

Brief Description of the Figures

Embodiments according to the present invention will subsequently be described taking reference to the enclosed figures in which:

au o s gna . It shou be noted that the nput audio signal 1 10 can be a single-channel audio signal but is preferably a multi-channel audio signal. Similarly, the processed audio signal 1 12 can be a single-channel audio signal but is preferably a multi-channel audio signal. The signal processor 100 may, for example, comprise a coefficient estimation block or coefficient estimation unit 120, which is configured to estimate coefficients 124 of an autoregressive reverberation model (for example, AR coefficients or MAR coefficients of a multi-channel autoregressive reverberation model) using the single-channel or multichannel input audio signal 1 10 and a delayed noise-reduced reverberant signal 122.

For example, the estimation of the coefficients of the autoregressive reverberation model 120 and may receive the input audio signal 1 10 and the delayed noise-reduced reverberant signal 122.

The signal processor 100 also comprises a noise reduction unit or noise reduction block 130 which receives the input audio signal 1 10 and which provides a noise-reduced (but typically reverberant or non-reverberation-reduced) signal 132. The noise reduction unit or noise reduction block 130 is configured to provide a noise-reduced (but typically reverberant) signal using the (typically noisy and reverberant) input audio signal 1 10 and the estimated coefficients 124 of the autoregressive reverberation model which are provided by the estimation block or estimation unit 120.

It should be noted here that the noise reduction 130 may, for example, use coefficients 124 of the autoregressive reverberation model which have been obtained on the basis of a previously determined noise-reduced reverberant signal 132 (possibly in combination with the input audio signal 1 10).

The apparatus 100 optionally comprises a delay block or delay unit 140, which may be configured to obtain the noise-reduced reverberant signal 132 provided by the noise reduction unit or noise reduction block 130 to provide, as an output, a delayed version 122 thereof. Accordingly, the estimation 120 of the coefficients of the autoregressive reverberation model can operate on a previously obtained (derived) noise-reduced reverberant signal (which is provided or derived by the noise reduction block 130) and the input audio signal 1 10.

The apparatus 100 also comprises a block or unit 150 for the derivation of a noise- reduced and reverberation-reduced output signal, which may serve as the processed audio signal 1 12. The block or unit 150 preferably receives the noise-reduced reverberant signal 132 from the noise reduction block or noise reduction unit 130 and the coefficients 124 of the autoregressive reverberation model provided by the estimation block or estimation unit 120. Thus, the block or unit 150 may, for example, remove or reduce reverberation from the noise-reduced reverberant signal 132. For example, an appropriate filtering, in combination with a cancellation operation (for example, in a spectral domain) may be used for this purpose, wherein the coefficients 124 of the autoregressive reverberation model may determine the filtering (which is used to estimate the reverberation).

Regarding the apparatus 100, it should be noted that the separation of functionalities into blocks or units can be considered as an efficient but arbitrary choice. The functionalities described herein could also be distributed differently to a hardware apparatus as long as the fundamental functionality is maintained. Also, it should be noted that the blocks or units could be software blocks or software units which reuse the same hardware (like, for example, a microprocessor).

Regarding the functionality of the apparatus 100, it can be said that the separation between the noise reduction functionality (noise reduction block or noise reduction unit 130) and the estimation of the coefficients of the autoregressive reverberation model (estimation block or estimation unit 120) provides for a reasonably small computational complexity and still allows for obtaining a sufficiently good audio quality. Even though, theoretically, it would be best to estimate the noise-reduced and reverberation-reduced output signal using a joint cost function, it has been found that separately performing the noise reduction and the estimation of the coefficients of the autoregressive reverberation model using separate cost functions can still provide reasonably good results, while complexity can be reduced and stability problems can be avoided. Also, it has been found that the noise-reduced reverberant signal 132 serves as a very good intermediate quality, since the noise-reduced and reverberation-reduced output signal (i.e., the processed audio signal 1 12) can be derived from the noise-reduced (but reverberant or non- reverberation-reduced) signal 132 with little effort provided that the coefficients 124 of the autoregressive reverberation model are known.

However, it should be noted that the apparatus 100 as described in Fig. 1 can be supplemented by any of the features, functionalities and details described in the following, both individually and taken in combination.

2. Embodiments According to Figs. 3, 4 and 5 In the following, some additional embodiments will be described taking reference to Figs. 3, 4 and 5. However, before details of the embodiments will be described, some information regarding conventional solutions will be described and a signal model will be defined.

Generally speaking, methods and apparatuses for online dereverberation and noise reduction (using a parallel structure), optionally with reduction control, will be described.

2.1 Introduction The following embodiments of the invention are in the field of acoustic field processing, for example to remove reverberation noise from one or multiple microphones.

In distant speech communication scenarios, where the desired speech source is far from the capturing device, the speech quality and intelligibility as well as the performance of speech recognizers is typically degraded due to high levels of reverberation and noise compared to the desired speech level.

Dereverberation methods based on an autoregressive (AR) model per frequency band in the short-time Fourier transform (STFT) domain have been shown to perform superior to other reverberation models. Dereverberation methods based on this model typically solve the problem using approaches related to linear prediction. Furthermore, the general multichannel autoregressive (MAR) model is valid for multiple sources and can be formulated such that it provides the same number of channels at the output as at the input. Since the resulting enhancement process, which is a linear filter per frequency band across multiple STFT frames, does not change the spatial correlation of the desired signal, the enhancement is suitable as preprocessing for further array processing techniques.

While most existing techniques based on the MAR model are batch algorithms [Nakatani 2010, Yoshioka 2009, Yoshioka 2012], some online algorithms have been proposed in [Yoshioka 2013, Togami 2019, Jukic 2016]. However, the challenging problem in noisy environments using an online algorithm has only been addressed in [Togami 2015].

It has been found that, in noisy environments, the problem can be typically be solved by first performing a noise reduction step, followed by linear prediction-based methods to estimate the MAR coefficients (also known as room regression coefficients) and then filtering the signal.

Methods to estimate the variables x(k, n) and c(n) in a batch algorithm, where the coefficients c(n) are assumed stationary are proposed in [Yoshioka2009, Togami2013]. However, it has been found that in common realistic applications, the acoustic scene, i.e., the MAR coefficients <¾/?) , can be time-varying. The only online solution to the MAR coefficient estimation problem in noisy environments is proposed in [Togami2015], although under the assumption that the MAR coefficients are stationary.

Conventional approaches for such similar problems to estimate an AR signal and the AR parameters use a sequential structure as shown in Fig. 2, such as the conventional online approach [Togami2015]. First, a noise reduction stage 202 tries to remove the noise from the observed signals y(n) , and in a second step 203 the AR coefficients c(n) are estimated from the output signals of the first stage X(fj). It has been found that this structure is suboptimal for two reasons: 1 ) The MAR parameter estimation stage 203 assumes that the estimated signal x(n) is noise-free, which is often not possible in practice. 2) To use the information of the MAR coefficients in the noise reduction stage 202, the coefficients have to be assumed stationary, as the assumption c(n) = c(n - l ) is required to feed the estimated MAR coefficients back from the MAR coefficient estimation stage to the noise reduction stage.

To conclude, Fig. 2 shows a block schematic diagram of a conventional structure for MAR coefficient estimation in a noisy environment. The apparatus 200 comprises a noise statistics estimation 201 , a noise reduction 202, an AR coefficient estimation 203 and a reverberation estimation 204.

In other words, blocks 201 to 204 are blocks of the conventional sequential noise reduction and the reverberation system.

2.3 Embodiments According to the Present Invention

In the following, three embodiments according to the present invention will be described. Fig. 3 shows a block schematic diagram of embodiment 2 according to the present invention. Fig. 4 shows a block schematic diagram of embodiment 3 according to the present invention. Fig. 5 shows a block schematic diagram of embodiment 4 according to the present invention.

The apparatus 300 also comprises an autoregressive coefficient estimation 302 (AR coefficient estimation) which is configured to receive the input audio signal 301 and a delayed version (or past version) of the noise-reduced (but typically reverberant) signal 303a provided by the noise reduction 303. Moreover, the autoregressive coefficient estimation 302 is configured to provide the coefficients 302a of the autoregressive reverberation model.

The apparatus 300 optionally comprises a delayer 320 which is configured to derive the delayed version 320a from the noise-reduced (but typically reverberant) signal 303a provided by the noise reduction 303.

The apparatus 300 also comprises a reverberation estimation 304, which is configured to receive the delayed version 320a of the noise-reduced (but typically reverberant) signal 303a provided by the noise reduction 303. Moreover, the reverberation estimation 304 also receives the coefficients 302a of the autoregressive reverberation model from the autoregressive coefficient estimation 302. The reverberation estimation 304 provides an estimated reverberation signal 304a. The apparatus 300 also comprises a signal subtractor 330 which is configured to remove (or subtract) the estimated reverberation signal 304a from the noise-reduced (but typically reverberant) signal 303a provided by the noise reduction 303, to thereby obtain the processed audio signal 312, which is typically noise-reduced and reverberation-reduced. In the following, the functionality of the apparatus 300 according to Fig. 3 will be described in more detail. In particular, it should be noted that the autoregressive coefficient estimation 302 uses both the input signal 310 and the noise-reduced (but typically reverberant) output signal 303a of the noise reduction 303 (or, more precisely, a delayed version 320a thereof). Accordingly, the autoregressive coefficient estimation 302 can be performed separately from the noise reduction 303, wherein the noise reduction 303 can nevertheless take benefit of the coefficients 302a of the autoregressive reverberation model, and wherein the autoregressive coefficient estimation 302 can nevertheless take benefit of the noise-reduced signal 303a provided by the noise reduction 303. The reverberation can finally be removed from the noise-reduced (but typically reverberant) signal 303a provided by the noise reduction 303.

The apparatus or signal processor 500 according to Fig. 5 is similar to the apparatus or signal processor 400 according to Fig. 4, such that reference is made to the above explanations and such that equal components will not be described again.

However, the apparatus 500 also comprises a reverberation shaping 305 which receives the reverberation signal 304a provided by the reverberation estimation. The reverberation shaping 305 provides a shaped reverberation signal 305a. According to the concept as shown in Fig. 5, the reverberation signal 304a is subtracted from the sum of the scaled noise reduced signal 303b and the scaled input signal 410a. accordingly, an intermediate signal 520 is obtained. Moreover, a scaled version 305b of the shaped reverberation signal 305a is added to the intermediate signal 520 in order to obtain an output signal 512.

However, a direct combination of the signals 410a, 303b, 304a and 305b would be possible as well (without using an intermediate signal).

Accordingly, the apparatus 500 allows to adjust characteristics of the output signal 512. The original reverberation can be removed (at least to a large degree), for example by subtracting the (estimated) reverberation signal 304a from the sum of signals 303b, 410a. Accordingly, a modified (shaped) reverberation signal 305b can be added (for example after an optional scaling), to thereby obtain the output signal 512. Accordingly, the output signal can be obtained with a shaped reverberation and with an adjustable degree of noise reduction.

In the following, the embodiment according to Figs. 4 and 5, Fig. 5 will be summarized in other words.

The parallel structure shown in Fig. 3 (with some extensions and amendments) allows for an easy and effective way to control the amount of reverberation and noise reduction. Such a control can be desired in speech communication scenarios to keep e.g., some residual noise and reverberation for perceptual reasons or to mask artifacts produced by the reduction algorithm. We define the (desired) new output signal z(n) = s(n)+#r(n)+A,v(A7), where β_Γ and are the control parameters for the residual reverberation and noise. By re-arranging the equation and replacing unknown variables by the available estimates, we can compute the controlled output signals (e.g., the output signal (412) by z(n) = /?„y(n) + (1 - β_ν)χίη) )r (n) as shown in Fig. 4. The processing Blocks 301 and 302 are omitted in this Fig. 4 (but can optionally be added).

For further spectral and dynamic shaping of the residual reverberation, an optional processing of the reverberation signal f(f?) can be inserted as shown in Fig. 4 in Block

305 (for example, as shown in Fig. 5). The output signal with reverberation shaping is then computed by - β_νγ(")+ (1 - β_ν)Ηη)- Ηη)+β_Γ ), where r_s(n) is the shaped reverberation signal by Block 305. The reverberation shaping can be performed for example by an equalizer or compressor / expander commonly used in audio and music production.

3. Embodiments According to Figs. 7 and 9

In the following, further embodiments for a linear-prediction based online dereverberation and noise reduction using alternating Kalman filters will be described.

For example, Linear Prediction Based Online Dereverberation and Noise Reduction Using Alternating Kalman Filters will be described.

3.1 Introduction and Overview In the following, an overview of the concept underlying embodiments according to the present invention will be described. Multi-channel linear prediction based dereverberation in the short-time Fourier transform (STFT) domain has been shown to be highly effective. However, it has been found that to use such methods in the presence of noise, especially in the case of online processing, remains a challenging problem. To address this problem, an alternating minimization algorithm that consists of two interactive Kalman filters to estimate the noise-free reverberant signal and the multi-channel autoregressive (MAR) coefficients is proposed. The desired dereverberated signals are then obtained by filtering the noise-free signals (or noise-reduced signals) using the estimated MAR coefficients.

It has been found that existing sequential enhancement structures used for similar problems have a causality issue that both the optimal noise reduction and the reverberation stages depend on the current output of each other. To overcome this causality problem, a novel parallel dual Kalman structure is developed, which solves the problem using alternating Kalman filters. It has been found that this causality is important when dealing with time-variant acoustic scenarios, where the MAR coefficients are non- stationary.

The proposed method is evaluated using simulated and measured acoustic impulse responses and compared to a method based on the same signal model. In addition, a method (and concept) to control the amount of reverberation and noise reduction independently is described.

To conclude, embodiments according to the invention can be used for a dereverberation. Embodiments according to the invention use a multi-channel linear prediction and an autoregressive model. Embodiments according to the invention use a Kalman filter, preferably in combination with an alternating minimization.

In the present application (and, in particular, in this section) a method (and concept) based on the MAR reverberation model is proposed to reduce reverberation and noise using an online algorithm. The proposed solution outperforms the noise-free solution presented in [3] where the MAR coefficients are modeled by a time-varying first-order Markov model. To obtain the desired dereverberated speech signals, it is possible to estimate the MAR coefficients and the noise-free reverberant speech signal.

The proposed solution has several advantages to conventional solutions: Firstly in contrast to the sequential signal and autoregressive (AR) parameter estimation methods used for noise reductions presented in [8] and [17], a parallel estimation structure as an alternating minimization algorithm using, for example, two interactive Kalman filters to estimate the MAR coefficients and the noise-free reverberant signals is proposed. This parallel structure allows a fully causal estimation chain as opposed to a sequential structure, where the noise reduction stage would use outdated MAR coefficients.

Secondly, in the proposed method we (optionally) assume a randomly time-varying MAR process instead of computing a time-invariant linear filter and a time-varying non-linear filter like in an expectation-maximization (EM) algorithm proposed in [31 ]. Thirdly, the proposed algorithm and concept does not require multiple iterations per time frame but can be an adaptive algorithm that converges over time. Finally, as an optional extension, a method to control the amount of reverberation and noise reduction independently is also proposed. The remainder of this section is organized as follows:

In subsection 2, the signal models for the reverberant signal, the noisy observation and the MAR coefficients are presented and the problem is formulated. In subsection 3, two alternating Kalman filters are derived as part of an alternating minimization problem to estimate the MAR coefficients and the noise-free signals. An optional method to control the reverberation and noise reduction is presented in subsection 4. In subsection 5, the proposed method and concept is evaluated and compared to state-of-the-art methods. Some conclusions are presented in subsection 6. Regarding the notation, it should be noted that factors are denoted as lower case bold symbols, for example a. Matrices are denoted as upper case bold symbols, for example A and scalars in normal font (e.g. , A). Estimated quantities are denoted by ^", for example A .

In the embodiments, estimated quantities may optionally take the place of ideal quantities.

3,2 Signal Model and Problem Formulation

respectively, where the ML x ML propagation matrix F(n) contains the MAR coefficients the filter may be time-varying, wherein it is assumed that a previous set of filter coefficients is scaled by a matrix A and affected by a "process noise" w(n).

Furthermore, in the signal model of y(n) is assumed that the background noise signal v(n) is added to the reverberant signal x(n).

However, it should be noted that the generative model of the reverberant signal, of the multi-channel autoregressive coefficients and of the noisy observation as shown in Fig. 6 should be considered as the example only.

D. Problem formulation

Our goal is to obtain an estimate of the early speech signals s(?i). Instead of directly estimating s(n), we propose to first estimate the noise-free reverberant signals x(n) and the MAR coefficients c(n), denoted by x(n) and c(n). Then we can obtain an estimate of

B Reference methods (optional)

To show the effectiveness and performance of the proposed method (dual-Kalman), we compare it to the following two methods:

^• single-Kalman: A single Kalman filter to estimate the MAR coefficients without noise reduction as proposed in [3]. The original algorithm assumes no additive noise. However, it can be still used to estimate the MAR coefficients from the noisy signal and then obtain a dereverberated, but still noisy filtered signal as output.

^• MAP-EM: In the method proposed in [31], the MAR coefficients are estimated using a Bayesian approach based on MAP estimation and the noise-free desired signal is then estimated using an EM algorithm. The algorithm is online, but the EM procedure requires about 20 iterations per frame to converge.

C. Results

1 ) Dependence on number of microphones: We investigated the performance of the proposed algorithm depending on the number of microphones M. The desired signal with a total length of 34 s consisted of two non-concurrent speakers at different positions: During the first 15 s the first speaker was active, while after 1 5 s, the second speaker was active. Each speaker signal was concolved with measured RIRs at different positions with with a T₆₀ = 630 ms. Stationary pink noise was added to the reverberant signals with iSNR = 15 dB. Figure 10 shows CD, PESQ, SIR and SRMR for a varying number of microphones M . The measures for the noisy reverberant input signal are indicated as light grey dashed line, and the SRMR of the target signal, i. e. the early speech, is indicated as dark grey dash-dotted line. For = 1, the CD is larger than for the input signal, which indicates an overall quality deterioration, whereas PESQ, SIR and SRMR still improve over the input, i. e. reverberation and noise are reduced. The performance in terms of all measures increases by increasing the number of microphones.

2) Dependence on filter length

The effect of the filter length L was investigated using measured RIR with different reverberation times. As in the first experiment, two non-concurrent speakers were active at different positions, and stationary pink noise was added with iSN R = 15 dB. Figure 1 1

nowe ge o t e orace esre sgna vector s(n), we can compute t e rever eraton Embodiments according to the invention can optionally comprise one or more of the following features:

• Receiving at least one microphone signal, or, alternatively, receiving at least two

microphone signals (optional).

• Transforming the microphone signal or the microphone signals into the time-frequency domain or another suitable domain (optional).

• Estimating the noise co variance matrix (optional).

• Using a parallel estimation structure for joint estimation of MAR coefficients and noise- free reverberant signal.

• The MAR coefficients are estimated using the noisy reverberant input signals and delayed estimated reverberant output signals from the noise reduction stage.

• The noise reduction stage receives current MAR coefficient estimates in each frame (optional).

· Computing the output signal (or, alternatively, output signals) by filtering the noise-free reverberant signal (or, alternatively, noise-free reverberant signals) (optional).

• Computing a controlled output signal (or, alternatively, output signals) from the

estimated signal components to set the amount of residual noise and reverberation (optional).

· Optionally computing a modified output signal (or, alternately, output signals) by

adding one or more processed/shaped reverberation signals with a certain level to the estimated dereverberated signal (or, alternately, estimated dereverberated signals) to achieve a different reverberation characteristic at the output signal.

To further conclude, in the present description, different inventive embodiments and aspects have been described in a chapter "Method and Apparatus for Dereverberation and Noise Reduction (using a parallel structure) With Reduction Control" (Section 2) and in a chapter "Linear Prediction Based Online Dereverberation and Noise Reduction Using Alternating Kalman Filters" (Section 3).

Also, further embodiments are defined by the enclosed claims and in the other sections (e.g. in the section "Summary of the invention" and in Section 1 .) It should be noted that any embodiment as defined by the claims can be supplemented by any of the details (for example, features and functionalities) described herein. Also, the embodiments described in the above mentioned sections can be used individually and can also be supplemented by any of the features in another section or by any feature included in the claims. Also, it should be noted that the individual aspects described herein can be used individually or in combination. Thus, details can be added to each of said individual aspects without adding details to another of the aspects.

It should also be noted that the present disclosure describes, explicitly or implicitly, features usable in an audio encoder (apparatus for providing an encoded representation of an input audio signal) and in an audio decoder (apparatus for providing a decoded representation of an audio signal on the basis of an encoded representation). Thus, any of the features described herein can be used in the context of an audio encoder and in the context of an audio decoder.

Moreover, features and functionalities disclosed herein relating to a method can also be used in an apparatus (configured to perform such a method or functionality). Furthermore, any of the features and functionalities disclosed herein with respect to an apparatus can also be used in a corresponding method. In other words, the methods disclosed herein can be supplemented by any of the features and functionalities described with respect to the apparatuses and vice versa. Also, any of the features and functionalities described herein can be implemented in hardware and software (or using hardware and/or software), or even a combination of hardware and software, as will be described in the section "Implementation Alternatives".

Also, it should be noted that the processing described herein may be performed, for example (but not necessarily) per frequency band or per frequency bin or for different frequency regions. It should be noted that aspects of the invention relate to a method and apparatus for online dereverberation and noise reduction with reduction control.

Embodiments according to the invention create a novel parallel structure for joint dereverberation and noise reduction. The reverberant signal is modelled, for example, using a narrowband multichannel autoregressive reverberation model with time-varying coefficients, which account for non-stationary acoustic environments. In contrast to existing sequential estimation structures, embodiments according to the invention estimate the noise-free reverberant signal and the autoregressive room coefficients in parallel, such that assumptions on stationary room coefficients are not required. In addition, a method to independently control the reduction level of noise and reverberation is proposed.

5. Method According to Fig. 14

Fig. 14 shows a flow chart of a method 1400 according to an embodiment of the present invention.

The method 1400 for providing a processed audio signal on the basis of an input audio signal comprises estimating 1410 coefficients of an autoregressive reverberation model using the input audio signal and a delayed noise-reduced reverberant signal obtained using a noise reduction stage.

The method also comprises providing 1420 a noise-reduced reverberant signal using the input audio signal and the estimated coefficients of the autoregressive reverberation model.

The method also comprises deriving 1430 a noise-reduced and reverberation-reduced output signal using the noise-reduced reverberant signal and the estimated coefficients of the autoregressive reverberation model.

The method 1400 can optionally be supplemented by any of the features, functionalities and details describer herein, both individually and in combination.

6. Implem e n tat ion a I te rna t j ye s

Although some aspects have been described in the context of an apparatus, it is clear that these aspects also represent a description of the corresponding method, where a block or device corresponds to a method step or a feature of a method step. Analogously, aspects described in the context of a method step also represent a description of a corresponding block or item or feature of a corresponding apparatus. Some or all of the method steps may be executed by (or using) a hardware apparatus, like for example, a microprocessor, a programmable computer or an electronic circuit. In some embodiments, one or more of the most important method steps may be executed by such an apparatus.

Depending on certain implementation requirements, embodiments of the invention can be implemented in hardware or in software. The implementation can be performed using a digital storage medium, for example a floppy disk, a DVD, a Blu-Ray, a CD, a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable control signals stored thereon, which cooperate (or are capable of cooperating) with a programmable computer system such that the respective method is performed. Therefore, the digital storage medium may be computer readable.

Some embodiments according to the invention comprise a data carrier having electronically readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods described herein is performed.

Generally, embodiments of the present invention can be implemented as a computer program product with a program code, the program code being operative for performing one of the methods when the computer program product runs on a computer. The program code may for example be stored on a machine readable carrier.

Other embodiments comprise the computer program for performing one of the methods described herein, stored on a machine readable carrier. In other words, an embodiment of the inventive method is, therefore, a computer program having a program code for performing one of the methods described herein, when the computer program runs on a computer.

A further embodiment of the inventive methods is, therefore, a data carrier (or a digital storage medium, or a computer-readable medium) comprising, recorded thereon, the computer program for performing one of the methods described herein. The data carrier, the digital storage medium or the recorded medium are typically tangible and/or non- transitionary. A further embodiment of the inventive method is, therefore, a data stream or a sequence of signals representing the computer program for performing one of the methods described herein. The data stream or the sequence of signals may for example be configured to be transferred via a data communication connection, for example via the Internet. A further embodiment comprises a processing means, for example a computer, or a programmable logic device, configured to or adapted to perform one of the methods described herein.

A further embodiment comprises a computer having installed thereon the computer program for performing one of the methods described herein.

A further embodiment according to the invention comprises an apparatus or a system configured to transfer (for example, electronically or optically) a computer program for performing one of the methods described herein to a receiver. The receiver may, for example, be a computer, a mobile device, a memory device or the like. The apparatus or system may, for example, comprise a file server for transferring the computer program to the receiver.

In some embodiments, a programmable logic device (for example a field programmable gate array) may be used to perform some or all of the functionalities of the methods described herein. In some embodiments, a field programmable gate array may cooperate with a microprocessor in order to perform one of the methods described herein. Generally, the methods are preferably performed by any hardware apparatus. The apparatus described herein may be implemented using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer.

The apparatus described herein, or any components of the apparatus described herein, may be implemented at least partially in hardware and/or in software.

The methods described herein may be performed using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer.

The methods described herein, or any components of the apparatus described herein, may be performed at least partially by hardware and/or by software. The above described embodiments are merely illustrative for the principles of the present invention. It is understood that modifications and variations of the arrangements and the details described herein will be apparent to others skilled in the art. It is the intent, therefore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of description and explanation of the embodiments herein.

[7] Y. Ephraim and D. Ma!ah, "Speech enhancement using a minimum-mean square error short-time spectral amplitude estimator," IEEE Trans. Acoust.. Speech, Signal Process., vol. 32, no. 6, pp. 1 109-1 121 , Dec. 1984.

[8] S. Gannot, D. Burshtein, and E. Weinstein, "Iterative and sequential Kalman filter- based speech enhancement algorithms," IEEE Trans. Speech Audio Process. , vol. 6, no. 4, pp. 373-385, Jul. 1998. [9] T. Gerkmann and R. C. Hendriks, "Unbiased MMSE-based noise power estimation with low complexity and low tracking delay," IEEE Trans. Audio, Speech, Lang. Process., vol. 20, no. 4, pp. 1383 -1393, May 2012.

[10] S. Goetze, A. Warzybok, I. Kodrasi, J. O. Jungmann, B. Cauchi, J. Rennies, E. A. P. Habets, A. Mertins, T. Gerkmann, S. Doclo, and B. Kollmeier, "A study on speech quality and speech intelligibility measures for quality assessment of single-channel dereverberation algorithms," in Proc. Intl. Workshop Acoust. Signal Enhancement (IWAENC), Sep. 2014, pp. 233-237. [1 1] ITU-T, Perceptual evaluation of speech quality (PESQ), an objective method for end- to-end speech quality assessment of narrowband telephone networks and speech codecs, International Telecommunications Union (ITU-T) Recommendation P.862, Feb. 2001 .

[12] A. Jukic, Z. Wang, T. van Waterschoot, T. Gerkmann, and S. Doclo, "Constrained multi-channel linear prediction for adaptive speech dereverberation," in Proc. Intl. Workshop Acoust. Signal Enhancement (IWAENC), Xi'an, China, Sep. 2016.

[13] A. Jukic, T. van Waterschoot, and S. Doclo, "Adaptive speech dereverberation using constrained sparse multichannel linear prediction," IEEE Signal Process. Lett. , vol. 24, no. 1 , pp. 101-105, Jan 2017.

[14] R. E. Kalman, "A new approach to linear filtering and prediction problems," Trans, of the AS ME Journal of Basic Engineering, vol. 82, no. Series D, pp. 35-45, 1960. [15] K. Kinoshita, M. Delcroix, S. Gannot, E. A. P. Habets, R. Haeb-Umbach, W. Kellermann, V. Leutnant, R. Maas, T. Nakatani, B. Raj, A. Sehr, and T. Yoshioka, "A summary of the REVERB challenge: state-of-the-art and remaining challenges in reverberant speech processing research," EURASIP Journal on Advances in Signal Processing, vol. 2016, no. 1 , p. 7, Jan 2016. [16] N. Kitawaki, H. Nagabuchi, and K. Itoh, Objective quality evaluation for low bit-rate speech coding systems," IEEE J. Sel. Areas Commun., vol. 6, no. 2, pp. 262-273, 1988.

[17] D. Labarre, E. Grivel, Y. Berthoumieu, E. Todini, and M. Najim, "Consistent estimation of autoregressive parameters from noisy observations based on two interacting Kalman filters," Signal Processing, vol. 86, no. 10, pp. 2863 - 2876, 2006, special Section: Fractional Calculus Applications in Signals and Systems.

[18] P. C. Loizou, Speech Enhancement Theory and Practice. 1 em plus 0.5em minus 0.4em Taylor & Francis, 2007.

[19] R. Martin, "Noise power spectral density estimation based on optimal smoothing and minimum statistics," IEEE Trans. Speech Audio Process., vol. 9, pp. 504-512, Jul. 2001.

[20] M. Miyoshi and Y. Kaneda, "Inverse filtering of room acoustics," IEEE Trans. Acoust., Speech, Signal Process., vol. 36, no. 2, pp. 145-152, Feb. 1988.

[21 ] T. Nakatani, T. Yoshioka, K. Kinoshita, M. Miyoshi, and J. Biing-Hwang, "Speech dereverberation based on variance-normalized delayed linear prediction," IEEE Trans. Audio, Speech, Lang. Process., vol. 18, no. 7, pp. 1717-1731 , 2010.

[22] P. A. Naylor and N. D. Gaubitch, Eds., Speech Dereverberation. 1 em plus 0.5em minus 0.4em London, UK: Springer, 2010.

[23] U. Niesen, D. Shah, and G. W. Wornell, "Adaptive alternating minimization algorithms," IEEE Transactions on Information Theory, vol. 55, no. 3, pp. 1423-1429, March 2009.

[24] J. F. Santos, M. Senoussaoui, and T. H. Falk, "An updated objective intelligibility estimation metric for normal hearing listeners under noise and reverberation," in Proc. Intl. Workshop Acoust. Signal Enhancement (IWAENC), Antibes, France, Sep. 2014. [34] T. Yoshioka, A. Sehr, M. Delcroix, K. Kinoshita, R. Maas, T. Nakatani, and W. Kellermann, "Making machines understand us in reverberant rooms: Robustness against reverberation for automatic speech recognition," IEEE Signal Processing Magazine, vol. 29, no. 6, pp. 1 14-126, Nov 2012.

[35] T. Yoshioka and T. Nakatani, "Dereverberation for reverberation-robust microphone arrays," in Proc. European Signal Processing Conf. (EUSIPCO), Sept 2013, pp. 1-5.

[36] [Online]. Available: http://www.audiolabs- erlangen.de/fau/professor/habets/software/signal-generator

Claims

Claims

1. A signal processor (100;300;400;500; 700;900) for providing one or more processed audio signals (1 12; 312;412;512; s(n); z(n)) on the basis of one or more input audio signals (1 10;310;410;710;910;y(n)). wherein the signal processor is configured to estimate coefficients (c (n)) of an autoregressive reverberation model using the one or more input audio signals and one or more delayed noise-reduced reverberant signals (x(n)) obtained using a noise reduction (130;303;703;903); and wherein the signal processor is configured to provide one or more noise-reduced reverberant signals (x(n)) using the input audio signal and the estimated coefficients (124;302a;702a; (n)) of the autoregressive reverberation model; and wherein the signal processor is configured to derive one or more noise-reduced and reverberation-reduced output signals (1 12; 312; 412; 512; s(n); z(n)) using the one or more noise-reduced reverberant signals (x(n)) and the estimated coefficients ( (n)) of the autoregressive reverberation model.

2. The signal processor ( 00;300;400;500; 700;900) according to claim , wherein the signal processor is configured to estimate coefficients ( (n)) of a multichannel autoregressive reverberation model.

3. The signal processor (100;300;400;500; 700;900) according to one of claims 1 to 2, wherein the signal processor is configured to use estimated coefficients ( (n)) of the autoregressive reverberation model associated with a currently processed portion of the input audio signal in order to provide the noise-reduced reverberant signal ( (n)) associated with the currently processed portion of the input audio signal

(1 10;310;410;710;910;y(n)).

4. The signal processor (100;300;400;500; 700;900) according to one of claims 1 to 3, wherein the signal processor is configured to use one or more delayed noise-reduced reverberant signals (x(n)) associated with a previously processed portion of the input audio signal ( 1 10;310;410;710;910;y(n)) for an estimation of coefficients (c(n)) of the autoregressive reverberation model associated with a currently processed portion of the input audio signal,

5. The signal processor (100;300;400;500; 700;900) according to one of claims 1 to 4, wherein the signal processor is configured to alternatingly provide estimated coefficients ( (n)) of the autoregressive reverberation model and noise-reduced reverberant signal portions (2(n)), and wherein the signal processor is configured to use estimated coefficients ( (n)) of the autoregressive reverberation model for the provision of the noise-reduced reverberant signal portions (x(n)), and wherein the signal processor is configured to use one or more delayed noise-reduced reverberant signals ( (n)) for the estimation of coefficients (c(n)) of the multichannel autoregressive reverberation model.

6. The signal processor (100;300;400;500; 700;900) according to one of claims 1 to 5, wherein the signal processor is configured to apply an algorithm which minimizes a cost function in order to estimate the coefficients (c(n)) of the autoregressive reverberation model.

7. The signal processor (100;300;400;500; 700;900) according to claim 6, wherein the cost function used for the estimation of the coefficients (c(n)) of the autoregressive reverberation model is an expectation value for a mean squared error of the coefficients ( (n)) of the autoregressive reverberation model.

8. The signal processor ( 100;300;400;500; 700;900) according to claim 6 or claim 7, wherein the signal processor is configured to apply the algorithm for the minimization of the cost function in order to estimate the coefficients ( (n)) of the autoregressive reverberation model under the assumption that the noise-reduced reverberant signal ( (n)) is fixed.

9. The signal processor ( 100;300;400;500; 700;900) according to one of claims 1 to 8, wherein the signal processor is configured to apply an algorithm for a minimization of a cost function in order to estimate the noise-reduced reverberant signal (X(n)).

10. The signal processor (100;300;400;500; 700;900) according to claim 9, wherein the cost function used for the estimation of the reverberant signal (x(n)) is an expectation value for a mean squared error of the reverberant signal (x(n) ).

1 1 . The signal processor (100;300;400;500; 700;900) according to claim 9 or claim 0, wherein the signal processor is configured to apply the algorithm for the minimization of the cost function in order to estimate the reverberant signal (x(n)) under the assumption that the coefficients (c(n)) of the autoregressive reverberation model are fixed.

12. The signal processor ( 100;300;400;500; 700;900) according to one of claims 1 to 1 1 , wherein the signal processor is configured to determine a reverberation component ( 124; 304a;704a;904a; (n)) on the basis of estimated coefficients (c(n)) of the autoregressive reverberation model and on the basis of one or more delayed noise-reduced reverberant signals (x(n)) associated with a previously processed portion of the input audio signal ( 1 10;310;410;710;910;y(n)), and wherein the signal processor is configured to cancel the reverberation component ( (n)) from the noise-reduced reverberant signal (i(n) ) associated with a currently processed portion of the input audio signal (1 10;3 0;410;710;9 0;y(n)), in order to obtain the noise- reduced and reverberation-reduced output signal ( 1 12; 312;412;512; s(n); z(n)).

13. The signal processor ( 100;300;400;500; 700;900) according to one of claims 1 to 12, wherein the signal processor is configured to perform a weighted combination of the input audio signal ( 1 10;310;410;710;910;y(n)) and of the noise-reduced reverberant signal (2(n)) and of a reverberation component, in order to obtain the noise-reduced and reverberation-reduced output signal ( 1 12; 312;412;512; s(n); z(n)).

14. The signal processor (100;300;400;500; 700;900) according to claim 13, wherein the signal processor is configured to also include a shaped version (305a, (n)) of the reverberation component (304a, (n)) in the weighted combination.

15. The signal processor ( 100;300;400;500; 700;900) according to one of claims 1 to 14, wherein the signal processor is configured to estimate a statistic (301a; 701a; Φ„(η)) of a noise component of the input audio signal.

16. The signal processor ( 100;300;400;500; 700;900) according to one of claims 1 to 15, wherein the signal processor is configured to estimate a statistic (301 a, 701 a, Φ„(η)) of a noise component of the input audio signal during a non-speech period.

17. The signal processor ( 100;300;400;500; 700;900) according to one of claims 1 to 16, wherein the signal processor is configured to estimate the coefficients (c(n)) of the autoregressive reverberation model using a Kalman filter.

18. The signal processor ( 100;300;400;500; 700;900) according to one of claims 1 to 17, wherein the signal processor is configured to estimate the coefficients ( (n)) of the autoregressive reverberation model on the basis of

an estimated error matrix Φ_Δ(:{η - 1) of a vector of coefficients (c(n-1 )) of the autoregressive reverberation model;

an estimated covariance <P_w(n) of an uncertainty noise of the vector of coefficients ( (n)) of the autoregressive reverberation model;

a previous vector of coefficients (c(n-1 )) of the autoregressive reverberation model; one or more delayed noise-reduced reverberant signals (2(n);

an estimated covariance Φ,, (n) associated with noisy but reverberation reduced signal components of the input audio signal;

- the input audio signal (y(n)). wherein the method comprises estimating (1410) coefficients (c (n)) of an autoregressive reverberation model using the one or more input audio signals and one or more delayed noise-reduced reverberant signals obtained using a noise reduction; and wherein the method comprises providing ( 1420) one or more noise-reduced reverberant signals (x(n)) using the one or more input audio signals and the estimated coefficients (c(n)) of the autoregressive reverberation model; and wherein the method comprises deriving (1430) one or more noise-reduced and reverberation-reduced output signals (S(n)) using the one or more noise-reduced reverberant signals (X(n)) and the estimated coefficients ( (n)) of the autoregressive reverberation model.

26. A computer program for performing the method according to claim 25 when the computer program runs on a computer.